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-rw-r--r--channels/chan_iax2.c15
-rw-r--r--include/asterisk/frame.h2
-rw-r--r--main/rtp.c44
3 files changed, 34 insertions, 27 deletions
diff --git a/channels/chan_iax2.c b/channels/chan_iax2.c
index 7dc60fc41..12c9c553d 100644
--- a/channels/chan_iax2.c
+++ b/channels/chan_iax2.c
@@ -3022,7 +3022,7 @@ static void __get_from_jb(const void *p)
/* create an interpolation frame */
af.frametype = AST_FRAME_VOICE;
af.subclass = pvt->voiceformat;
- af.samples = frame.ms * 8;
+ af.samples = frame.ms * (ast_format_rate(pvt->voiceformat) / 1000);
af.src = "IAX2 JB interpolation";
af.delivery = ast_tvadd(pvt->rxcore, ast_samp2tv(next, 1000));
af.offset = AST_FRIENDLY_OFFSET;
@@ -3094,7 +3094,7 @@ static int schedule_delivery(struct iax_frame *fr, int updatehistory, int fromtr
if(fr->af.frametype == AST_FRAME_VOICE) {
type = JB_TYPE_VOICE;
- len = ast_codec_get_samples(&fr->af) / 8;
+ len = ast_codec_get_samples(&fr->af) / (ast_format_rate(fr->af.subclass) / 1000);
} else if(fr->af.frametype == AST_FRAME_CNG) {
type = JB_TYPE_SILENCE;
}
@@ -4399,6 +4399,7 @@ static unsigned int calc_timestamp(struct chan_iax2_pvt *p, unsigned int ts, str
int voice = 0;
int genuine = 0;
int adjust;
+ int rate = ast_format_rate(f->subclass) / 1000;
struct timeval *delivery = NULL;
@@ -4466,7 +4467,7 @@ static unsigned int calc_timestamp(struct chan_iax2_pvt *p, unsigned int ts, str
p->offset = ast_tvadd(p->offset, ast_samp2tv(adjust, 10000));
if (!p->nextpred) {
- p->nextpred = ms; /*f->samples / 8;*/
+ p->nextpred = ms; /*f->samples / rate;*/
if (p->nextpred <= p->lastsent)
p->nextpred = p->lastsent + 3;
}
@@ -4485,11 +4486,11 @@ static unsigned int calc_timestamp(struct chan_iax2_pvt *p, unsigned int ts, str
ast_debug(1, "predicted timestamp skew (%u) > max (%u), using real ts instead.\n",
abs(ms - p->nextpred), MAX_TIMESTAMP_SKEW);
- if (f->samples >= 8) /* check to make sure we dont core dump */
+ if (f->samples >= rate) /* check to make sure we dont core dump */
{
- int diff = ms % (f->samples / 8);
+ int diff = ms % (f->samples / rate);
if (diff)
- ms += f->samples/8 - diff;
+ ms += f->samples/rate - diff;
}
p->nextpred = ms;
@@ -4521,7 +4522,7 @@ static unsigned int calc_timestamp(struct chan_iax2_pvt *p, unsigned int ts, str
}
p->lastsent = ms;
if (voice)
- p->nextpred = p->nextpred + f->samples / 8;
+ p->nextpred = p->nextpred + f->samples / rate;
return ms;
}
diff --git a/include/asterisk/frame.h b/include/asterisk/frame.h
index 18f8c18b6..042dd222c 100644
--- a/include/asterisk/frame.h
+++ b/include/asterisk/frame.h
@@ -149,7 +149,7 @@ struct ast_frame {
int subclass;
/*! Length of data */
int datalen;
- /*! Number of 8khz samples in this frame */
+ /*! Number of samples in this frame */
int samples;
/*! Was the data malloc'd? i.e. should we free it when we discard the frame? */
int mallocd;
diff --git a/main/rtp.c b/main/rtp.c
index 240c5b17b..d60cfaf0b 100644
--- a/main/rtp.c
+++ b/main/rtp.c
@@ -682,6 +682,11 @@ int ast_rtcp_fd(struct ast_rtp *rtp)
return -1;
}
+static int rtp_get_rate(int subclass)
+{
+ return (subclass == AST_FORMAT_G722) ? 8000 : ast_format_rate(subclass);
+}
+
unsigned int ast_rtcp_calc_interval(struct ast_rtp *rtp)
{
unsigned int interval;
@@ -911,10 +916,10 @@ static struct ast_frame *process_cisco_dtmf(struct ast_rtp *rtp, unsigned char *
}
} else if ((rtp->resp == resp) && !power) {
f = send_dtmf(rtp, AST_FRAME_DTMF_END);
- f->samples = rtp->dtmfsamples * 8;
+ f->samples = rtp->dtmfsamples * (rtp_get_rate(f->subclass) / 1000);
rtp->resp = 0;
} else if (rtp->resp == resp)
- rtp->dtmfsamples += 20 * 8;
+ rtp->dtmfsamples += 20 * (rtp_get_rate(f->subclass) / 1000);
rtp->dtmf_timeout = dtmftimeout;
return f;
}
@@ -995,7 +1000,7 @@ static struct ast_frame *process_rfc2833(struct ast_rtp *rtp, unsigned char *dat
if ((rtp->lastevent != seqno) && rtp->resp) {
rtp->dtmf_duration = new_duration;
f = send_dtmf(rtp, AST_FRAME_DTMF_END);
- f->len = ast_tvdiff_ms(ast_samp2tv(rtp->dtmf_duration, 8000), ast_tv(0, 0));
+ f->len = ast_tvdiff_ms(ast_samp2tv(rtp->dtmf_duration, rtp_get_rate(f->subclass)), ast_tv(0, 0));
rtp->resp = 0;
rtp->dtmf_duration = rtp->dtmf_timeout = 0;
}
@@ -1005,7 +1010,7 @@ static struct ast_frame *process_rfc2833(struct ast_rtp *rtp, unsigned char *dat
if (rtp->resp && rtp->resp != resp) {
/* Another digit already began. End it */
f = send_dtmf(rtp, AST_FRAME_DTMF_END);
- f->len = ast_tvdiff_ms(ast_samp2tv(rtp->dtmf_duration, 8000), ast_tv(0, 0));
+ f->len = ast_tvdiff_ms(ast_samp2tv(rtp->dtmf_duration, rtp_get_rate(f->subclass)), ast_tv(0, 0));
rtp->resp = 0;
rtp->dtmf_duration = rtp->dtmf_timeout = 0;
}
@@ -1322,14 +1327,15 @@ static void calc_rxstamp(struct timeval *tv, struct ast_rtp *rtp, unsigned int t
double d;
double dtv;
double prog;
-
+ int rate = rtp_get_rate(rtp->f.subclass);
+
if ((!rtp->rxcore.tv_sec && !rtp->rxcore.tv_usec) || mark) {
gettimeofday(&rtp->rxcore, NULL);
rtp->drxcore = (double) rtp->rxcore.tv_sec + (double) rtp->rxcore.tv_usec / 1000000;
/* map timestamp to a real time */
rtp->seedrxts = timestamp; /* Their RTP timestamp started with this */
- rtp->rxcore.tv_sec -= timestamp / 8000;
- rtp->rxcore.tv_usec -= (timestamp % 8000) * 125;
+ rtp->rxcore.tv_sec -= timestamp / rate;
+ rtp->rxcore.tv_usec -= (timestamp % rate) * 125;
/* Round to 0.1ms for nice, pretty timestamps */
rtp->rxcore.tv_usec -= rtp->rxcore.tv_usec % 100;
if (rtp->rxcore.tv_usec < 0) {
@@ -1341,13 +1347,13 @@ static void calc_rxstamp(struct timeval *tv, struct ast_rtp *rtp, unsigned int t
gettimeofday(&now,NULL);
/* rxcore is the mapping between the RTP timestamp and _our_ real time from gettimeofday() */
- tv->tv_sec = rtp->rxcore.tv_sec + timestamp / 8000;
- tv->tv_usec = rtp->rxcore.tv_usec + (timestamp % 8000) * 125;
+ tv->tv_sec = rtp->rxcore.tv_sec + timestamp / rate;
+ tv->tv_usec = rtp->rxcore.tv_usec + (timestamp % rate) * 125;
if (tv->tv_usec >= 1000000) {
tv->tv_usec -= 1000000;
tv->tv_sec += 1;
}
- prog = (double)((timestamp-rtp->seedrxts)/8000.);
+ prog = (double)((timestamp-rtp->seedrxts)/(float)(rate));
dtv = (double)rtp->drxcore + (double)(prog);
current_time = (double)now.tv_sec + (double)now.tv_usec/1000000;
transit = current_time - dtv;
@@ -1643,7 +1649,7 @@ struct ast_frame *ast_rtp_read(struct ast_rtp *rtp)
if (rtp->resp) {
struct ast_frame *f;
f = send_dtmf(rtp, AST_FRAME_DTMF_END);
- f->len = ast_tvdiff_ms(ast_samp2tv(rtp->dtmf_duration, 8000), ast_tv(0, 0));
+ f->len = ast_tvdiff_ms(ast_samp2tv(rtp->dtmf_duration, rtp_get_rate(f->subclass)), ast_tv(0, 0));
rtp->resp = 0;
rtp->dtmf_timeout = rtp->dtmf_duration = 0;
return f;
@@ -1665,7 +1671,7 @@ struct ast_frame *ast_rtp_read(struct ast_rtp *rtp)
calc_rxstamp(&rtp->f.delivery, rtp, timestamp, mark);
/* Add timing data to let ast_generic_bridge() put the frame into a jitterbuf */
ast_set_flag(&rtp->f, AST_FRFLAG_HAS_TIMING_INFO);
- rtp->f.ts = timestamp / 8;
+ rtp->f.ts = timestamp / (rtp_get_rate(rtp->f.subclass) / 1000);
rtp->f.len = rtp->f.samples / ( (ast_format_rate(rtp->f.subclass) == 16000) ? 16 : 8 );
} else if(rtp->f.subclass & AST_FORMAT_VIDEO_MASK) {
/* Video -- samples is # of samples vs. 90000 */
@@ -3067,6 +3073,11 @@ static int ast_rtp_raw_write(struct ast_rtp *rtp, struct ast_frame *f, int codec
unsigned int ms;
int pred;
int mark = 0;
+ int rate = rtp_get_rate(f->subclass) / 1000;
+
+ if (f->subclass == AST_FORMAT_G722) {
+ f->samples /= 2;
+ }
if (rtp->sending_digit) {
return 0;
@@ -3078,7 +3089,7 @@ static int ast_rtp_raw_write(struct ast_rtp *rtp, struct ast_frame *f, int codec
pred = rtp->lastts + f->samples;
/* Re-calculate last TS */
- rtp->lastts = rtp->lastts + ms * 8;
+ rtp->lastts = rtp->lastts + ms * rate;
if (ast_tvzero(f->delivery)) {
/* If this isn't an absolute delivery time, Check if it is close to our prediction,
and if so, go with our prediction */
@@ -3133,7 +3144,7 @@ static int ast_rtp_raw_write(struct ast_rtp *rtp, struct ast_frame *f, int codec
rtp->lastdigitts = rtp->lastts;
if (ast_test_flag(f, AST_FRFLAG_HAS_TIMING_INFO))
- rtp->lastts = f->ts * 8;
+ rtp->lastts = f->ts * rate;
/* Get a pointer to the header */
rtpheader = (unsigned char *)(f->data - hdrlen);
@@ -3297,11 +3308,6 @@ int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *_f)
}
while ((f = ast_smoother_read(rtp->smoother)) && (f->data)) {
- if (f->subclass == AST_FORMAT_G722) {
- /* G.722 is silllllllllllllly */
- f->samples /= 2;
- }
-
ast_rtp_raw_write(rtp, f, codec);
}
} else {
generated by cgit v1.2.3 (git 2.25.1) at 2024-06-01 22:49:59 +0000